audio compression msccomputerscience.com. the process of digitizing audio signals is called pcm pcm...

msccomputerscience.com

AUDIO COMPRESSION

msccomputerscience.com

• The process of digitizing audio signals is called PCM• PCM involves sampling audio signal at minimum rate

which is twice the max freq.• When bandwidth of communication channel is less

than that of signal, sampling rate is determined by the bandwidth of communication channel. Such a signal is called Band limited signal

msccomputerscience.com

• For speech signal: Fmax =10KHZ Minimum sampling rate =20Ksps Bits required= 12bpsFor Stereophonic speech =240kbps

• For general audio: Fmax =20KHZ Minimum sampling rate =40Ksps Bits required= 16bpsFor Stereophonic speech =1.28Mbps

msccomputerscience.com

• In multimedia application for audio, bandwidth of communication channel is less

• Therefore audio signal can either be sampled at lower rate (or)use compression algorithm.

• Though sampling audio signal at lower rate is simple, but has its disadvantages

Quality reduces due to loss of high frequency components of original signal

use of fewer bits per sample introduces higher level of quantization noise

msccomputerscience.com

Differential Pulse Code Modulation(DPCM)

• Differential Pulse Code Modulation(DPCM) is a derivative of standard PCM

• For most audio signals range of difference in amp. < range of actual

between successive samples sample amp. • Therefore if digitized difference signal is used to

encode ,only fewer bits are required in comparison with PCM with same sampling rate

msccomputerscience.com

msccomputerscience.com

DPCM Encoder

• Register : A temporary storage facility which hold the digitized sample of analog input signal

• DPCM,differential signal is computed by subtracting Ro (current content of register )from the new digitized sample , output by ADC

• The value in the register is then updated by adding the computed difference signal output by subtractor, before it is transmitted.

msccomputerscience.com

DPCM Decoder

• The decode operates by simply adding the received difference signal DPCM to computed signal held in the register.

• Ro = current content of register• R1 = new/updated content• DPCM saves just bit, for a standard pcm voice

signal• i.e if bitrate=64kbps, using DPCM bitrate

becomes 56kbps.

msccomputerscience.com

• The output of ADC is directly used thus accuracy of each computed difference signal(called residual) depends on accuracy of previous signal held in register.

• But we know ADC operations produces quantization errors, which effects accuracy of value held in register and in turn effect DPCM residual signal

• To overcome this, and to predict more accurate previous signal, we have other sophisticated technique.

msccomputerscience.com

Predictive DPCM signal encoder

• Here differential signal is computed by subtracting varying proportion of last 3 predicted values from digital value output by ADC

msccomputerscience.com

Third order predictive DPCM signal encoder and decoder

msccomputerscience.com

• Ex. If predictor co-efficient C1=0.5,c2=c3=0.25• Then R1=c1*0.5 R2=c2*0.25 R3=c3*0.25 All the shifted values is added and then

resulting sum is subtracted from digitized output by ADC(PCM)

msccomputerscience.com

• Later the content of R is transferred to R2 and R2 to R3, and the new predicted value is loaded to R ready for next sample to be processed.

• The decoder operates by adding proportions of last 3 computed PCM to DPCM

msccomputerscience.com

Adaptive Differential PCM(ADPCM)

• Depending on the amplitude , no. of bits used for difference signal can be varied . This principle is used in ADPCM.

• This results in saving bandwidth and improves quality

• The principle is same as DPCM, except a 8 order predictor is used

msccomputerscience.com

International Standards

• International Standards for ADPCM is ITU-T recommendation G721

• Another standard is ITU-T recommendation G722 provides better sound quality

• This is used for applications such as conferencing .

msccomputerscience.com

ADPCM subband encode and decode scheme

msccomputerscience.com

• In order to allow higher signal band width , before sampling the audio signal , it is first passed through 2 filters

• One filter passes only freq. 50HZ through 3.5 KHZ Second filter passes only freq. 3.5KHz through 7KHzThe input signal is divided into Lower subband signal

and Upper subband signal ,both are sampled and encoded independently by ADPCM

The two bit streams are then multiplexed.